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import os |
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import librosa |
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import torch |
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import numpy as np |
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from fairseq import checkpoint_utils |
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from tqdm import tqdm |
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import torch |
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def load_hubert_model(hps): |
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ckpt_path = hps.hubert_file |
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print("Load Hubert Model...") |
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models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task( |
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[ckpt_path], |
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suffix="", |
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) |
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model = models[0] |
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model.eval() |
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if torch.cuda.is_available(): |
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model = model.cuda() |
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return model |
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def get_hubert_content(hmodel, wav_16k_tensor): |
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feats = wav_16k_tensor |
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if feats.dim() == 2: |
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feats = feats.mean(-1) |
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assert feats.dim() == 1, feats.dim() |
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feats = feats.view(1, -1) |
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padding_mask = torch.BoolTensor(feats.shape).fill_(False) |
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inputs = { |
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"source": feats.to(wav_16k_tensor.device), |
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"padding_mask": padding_mask.to(wav_16k_tensor.device), |
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"output_layer": 9, |
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} |
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with torch.no_grad(): |
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logits = hmodel.extract_features(**inputs) |
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feats = hmodel.final_proj(logits[0]).squeeze(0) |
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return feats |
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def content_vector_encoder(model, audio_path, default_sampling_rate=16000): |
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""" |
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# content vector default sr: 16000 |
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""" |
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wav16k, sr = librosa.load(audio_path, sr=default_sampling_rate) |
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device = next(model.parameters()).device |
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wav16k = torch.from_numpy(wav16k).to(device) |
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content_feature = get_hubert_content(model, wav_16k_tensor=wav16k) |
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return content_feature.cpu().detach().numpy() |
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def repeat_expand_2d(content, target_len): |
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""" |
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content : [hubert_dim(256), src_len] |
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target: [hubert_dim(256), target_len] |
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""" |
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src_len = content.shape[-1] |
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target = torch.zeros([content.shape[0], target_len], dtype=torch.float).to( |
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content.device |
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) |
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temp = torch.arange(src_len + 1) * target_len / src_len |
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current_pos = 0 |
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for i in range(target_len): |
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if i < temp[current_pos + 1]: |
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target[:, i] = content[:, current_pos] |
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else: |
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current_pos += 1 |
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target[:, i] = content[:, current_pos] |
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return target |
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def get_mapped_features(raw_content_features, mapping_features): |
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""" |
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Content Vector: frameshift = 20ms, hop_size = 480 in 24k |
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Now it's only used for mapping to bigvgan's mels (sr = 24k, hop_size = 256, frameshift ~= 10.7 ms) |
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""" |
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source_hop = 480 |
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target_hop = 256 |
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factor = np.gcd(source_hop, target_hop) |
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source_hop //= factor |
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target_hop //= factor |
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print( |
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"Mapping source's {} frames => target's {} frames".format( |
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target_hop, source_hop |
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) |
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) |
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results = [] |
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for index, mapping_feat in enumerate(tqdm(mapping_features)): |
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target_len = len(mapping_feat) |
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raw_feats = raw_content_features[index][0].cpu().numpy().T |
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source_len, width = raw_feats.shape |
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const = source_len * source_hop // target_hop * target_hop |
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up_sampling_feats = np.repeat(raw_feats, source_hop, axis=0) |
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down_sampling_feats = np.average( |
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up_sampling_feats[:const].reshape(-1, target_hop, width), axis=1 |
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) |
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err = abs(target_len - len(down_sampling_feats)) |
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if err > 3: |
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print("index:", index) |
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print("mels:", mapping_feat.shape) |
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print("raw content vector:", raw_feats.shape) |
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print("up_sampling:", up_sampling_feats.shape) |
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print("down_sampling_feats:", down_sampling_feats.shape) |
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exit() |
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if len(down_sampling_feats) < target_len: |
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end = down_sampling_feats[-1][None, :].repeat(err, axis=0) |
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down_sampling_feats = np.concatenate([down_sampling_feats, end], axis=0) |
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feats = down_sampling_feats[:target_len] |
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results.append(feats) |
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return results |
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def extract_hubert_features_of_dataset(datasets, model, out_dir): |
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for utt in tqdm(datasets): |
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uid = utt["Uid"] |
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audio_path = utt["Path"] |
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content_vector_feature = content_vector_encoder(model, audio_path) |
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save_path = os.path.join(out_dir, uid + ".npy") |
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np.save(save_path, content_vector_feature) |
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